Voice over Internet Protocol (VoIP) has become an attractive alternative to the traditional land line service using a public switched telephone network (PSTN). VoIP is also referred to as Internet telephony, broadband telephony, or voice over broadband. Regardless of the term used, voice and/or video are digitized and transmitted over a packet switched network. Audio and video coders/decoders (codecs) may be employed to provide compression or to enable high fidelity stereo sound. To ensure real-time delivery of voice and video the packet data may be further encapsulated in a protocol that can provide a quality of service, e.g., real-time transport protocol (RTP) or secure real-time transport protocol (SRTP).
As with traditional PSTN services, phone calls must be set up when a number is dialed and torn down when a user hangs up the phone. In VoIP systems, various proprietary or open signaling protocols may be used, such as session initiation protocol (SIP), IP multimedia subsystem (IMS), H.323, or Skype. Before signaling can begin, however, the VoIP phone or terminal involved in the call needs to be provisioned. For example, when SIP is used, a SIP user agent (UA) is resident in each VoIP phone or device. The SIP UA needs certain information such as its phone number, its media access control (MAC) or IP address, the SIP server IP address, and the SIP domain name. In current VoIP systems, this information is provided to the SIP UA manually.